[Live-devel] Problem of new supportfor unicastRTSP/RTPstreaming from a single input source

[Ross Finlayson]

>"MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
>exceeds the client's buffer size (20000).  104 bytes of trailing data will
>be dropped!"

This refers to the size of the buffer that the final 'sink' object in the 
data chain provides for receiving data.  If this is a "FileSink", then the 
default buffer size is 20000, which apparently turns out to be a bit 
smaller than the largest MPEG-4 video frame in your data.

If you are using "openRTSP" to receive the data, then you can fix this by 
adding
         -b 30000
to its command line.  Otherwise, you can add
         30000
as the third argument to
         FileSink::createNew()
(see "liveMedia/include/FileSink.hh")

>I try to use "increaseReceiveBufferTo()" to increase socket's buffer

No, that does something different - it increases the size of the buffer 
that the kernel uses to hold incoming UDP (e.g., RTP) packets.

>BTW, I test multiple unicast RTSP server for MPEG2 Transport Stream, still
>getting same error "Missing sync byte!"

That's because I haven't fixed that bug yet.


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>